asterisk users - Feb 2005 - 4xHFC-s cards vs 1 quadbri HFC-4S card ?

Hi,

I wonder what makes the difference between inserting 4 HFC-S cards (cca. 120
EUR)  and using 1 QuadBRI card (approx. 700 EUR) ?

What makes such difference ?  Is it possible to do first configuration ?
With what drivers ? Is it stable ?

Thanks in advance,

regards,

Rob.

Hallo Robert Rozman
On Tue, 15 Feb 2005 13:54:48 +0100 you wrote:
> Hi,
> 
> I wonder what makes the difference between inserting 4 HFC-S cards (cca.
> 120 EUR)  and using 1 QuadBRI card (approx. 700 EUR) ?
> 
> What makes such difference ?  Is it possible to do first configuration ?
1st: 4Slots for the HFCs + 4NTs to supply the S0 bus + IRQ hassle
(you need a box with bios to force IRQs, one more slot for ethernet, time
to get all this up and running)

2nd: Yes (for testing)        No for production (IMHO)
> With what drivers ? Is it stable ?
1st: bristuff, kernel 2610
2nd: my testbox works that way
> 
> Thanks in advance,
> 
> regards,
> 
> Rob.
> 
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-- 
Tho/\/\as

> I wonder what makes the difference between inserting 4 HFC-S cards 
> (cca. 120
> EUR)  and using 1 QuadBRI card (approx. 700 EUR) ?
>
> What makes such difference ?  Is it possible to do first configuration 
> ?
> With what drivers ? Is it stable ?
1 HFC-S card - lots of interrupts
4 cards - interrupt havoc

1 QuadBRI - some interrupts, but not too many

Hi,

I have a problem when configuring Asteriskand SER,
using SER as a simple SIP gateway. SER connects to
another third party SIP server. I want to call a user
that is registered in the third party SIP server, from
asterisk. In order to achieve this, I defined a peer
in "sip.conf", as follows:


[sip4_out]
type=peer          
secret=asterisk1
username=asterisk1		
fromuser=asterisk1		
insecure=no
context=home
host=10.2.250.151
fromdomain=antero.ssf.pt
port=5070

Then, I defined an outbound rule in "extensions.conf":

exten => _77.,1,Dial(SIP/${EXTEN:2}@sip4_out,30,r)


At this point, I am able to dial out SIP addresses,
using my SER and a SIP client. My problem is that ALL
the addresses that I dial are formatted in the SIP
INVITE message with the IP address and port of my SIP
peer (sip4_out), in the TO: field.
For example, if I dial 77sip4@mi.pt from my SIP
client, Asterisk will format the TO: field as:
	To: <sip:sip4@10.2.250.151:5070>

Is there anything we can do in the configuration files
to make Asterisk format that field with the actual SIP
address that was dialed? That is:
	To: <sip:sip4@mi.pt>

Many thanks for your help,
Paulo